In the past, data-centric and telephony-centric communications networks have existed as separate domains. More recently, however, computer users are taking advantage of PC-to-phone technology. Many online service providers seek to offer users with the ability to place telephone calls from their computer devices, using the computer devices as the equivalent of telephone handsets. A frequently used protocol for providing advanced telephony services across the Internet is referred to as Session Initiation Protocol (SIP). SIP, which is currently a proposed standard of the Internet Engineering Task Force (IETF) (see RFC 2543), is a text-based protocol for initiating interactive communication sessions between users. SIP can be used to establish, change and terminate calls between one or more users in a network based on the Internet Protocol (IP).
Moreover, society is becoming increasingly mobile and telephone users strongly desire the ability to place and receive calls from myriad locations. In other words, the ability for a telephone call to “follow” the user, whether logged in from work, home, or a mobile location, is needed.
Conventional telephone and voice systems enable a telephone customer to automatically direct incoming calls to different phones based on a fixed time schedule. Other known systems merely notify an online user of an incoming call on the telephone line being used for the online session. In this context, “online user” refers to a user of a computer presently connected to a network such as the Internet via dial-up access. Unfortunately, users of these conventional systems cannot easily and route their incoming calls dynamically. The need exists to route a user's incoming phone call to wherever the user is currently online but also based on user preferences.
In light of the foregoing, further improvements are needed for intelligent routing of real-time communications using user context such as schedule and presence.